I went to the ‘Channel Partners’ conference in Chicago back in September 2011 and this question was being asked so many times it got me thinking that.

There is no single and clear explanation that people can use over and over again and be sure that the client understands what SIP trunks are.

Of course, a lot of clever people out there have no problems explaining what a SIP trunk is and their clients are making SIP calls all day long and reaping the benefits of this ever evolving technology.

But after hearing this question repeated over and over again not just at Channel Partners but other events such as ITEXPO and Enterprise.

Connect I thought that people in the Telco business could help me out in my quest.

So, I placed the question on a number of groups on Linkedin and received around 90 responses.

The remit was to be able to describe SIP trunks to a client; so some of the good but overly technical suggestions we put to one side and after a little more sifting we got them down to what we (at The SIP School) thought were the better ones.

Then we posted them all on our website for people to vote on them.

353 votes later and we’ve now decided to announce the winner!

With 19% of the votes (67 out of 353) the winning description is:

“SIP Trunking is a Voice over IP (VoIP) solution that uses a data network or internet connection to connect a business telephone system to the PSTN.”

Well done to Bruce McKay of Allstream OK, so we have a winner yet with 19% of the votes it wasn’t a runaway victory and I’m certain that some of you may not completely agree – but there it is.

The ‘challenge’ was meant to be informative as well as fun and now if you need to find that snappy explanation to help your clients understand this could be the phrase you can use to ensure no glassy eyes!.

Why not try it out next time you are speaking with people and let us know your reactions on the Linkedin group SIP Trunking – Trials and Tribulations.

But wait..!

I’ve got a funny feeling that we may have to ask this question again at some point because there are a couple of things that I want to add to wrap up this (completely) non-scientific exercise.

Firstly, the winning comment mentions connecting to the PSTN.

Well maybe this should be saying the PSTN and other VoIP enabled networks?.

Connectivity to the PSTN is important yet as people try to avoid the PSTN then what are they connecting to?.

Yes, it’s other branches, partner companies etc, over Data networks.

Also, a fair number of entries were clear in saying that the term ‘Trunk’ shouldn’t be used at all!.

I understand why they say this i.e. it’s linked to old technology etc. etc.

But actually it’s a word that everyone is used; it’s not going to disappear, so let’s make it work by explaining it clearly.

Finally I have to say that if the winning explanation or even your own still confuses your client, then try the one from Steve Barret (Portel AV Ltd in the UK).

“I spent ages recently trying to explain what SIP was all about.

In the end I told my client it was a ‘magic door’ he understood this and asked why I hadn’t said that in the first place!”.

And with those final words, that’s it until the next challenge – Watch this space!.

Graham Francis is the CEO of the SIP School and can be contacted at graham@thesipschool.com.
FollowGraham on Twitter.

 Case Study: Resort Hotel & Cabins Company Hybrid VoIP / Traditional Telephony Upgrade & Expansion

Introduction

A French Mediterranean beach resort hotel and cabins company¹ (hereafter, “The Customer”) was expanding and modernizing their total unit and existing units at their location over the Winter of 2007. A shift in their clientele showed increased business-related traffic, such as retreats, weekend meetings and tiger-team think-tank sessions. As a result, they were planning on a significant Internet connection to support these customers, as well as an expansion of phone system services. They concluded that Voice Over Internet Protocol (VOIP) technology was a logical possibility to reduce cable-plant management issues as well as long-term total cost of ownership (TOC).

The original configuration and buildings were:

• Two primary buildings of 40 and 30 units respectively. These were already wired for analog telephone service, which the customer did not wish to “rip and replace”.
• Eleven cabin-styled buildings of three units each. The cabins were also wired for analog, however, environment issues were forcing a complete rebuild of the units.

The planned additions as part of the project were:

• Nine more cabin-style buildings
• A third primary building of 30 units
• Approximately 16 more administration phones scattered across the facilities.
• Replace the trio of aging FAX machines with a less paper-intensive service.

The new construction would be wired with CAT6 cable to support future gigabit application. The total unit count would then be 160 units, with one phone per unit.

Customer requirements for the PBX and services included:

• 100% control of the system needed to be from the front desk as well as the duty manager’s office.
• Due to its geographic location, there needed to be as much remote service capability as possible to avoid customer problems associated with non-functional phones.

This was presented to the team as a “Request For Proposal” from the client, who was not fully certain this list of needs was achievable without an unjustifiably large capital expenditure. We proceeded to assess the technologies available and provide the customer with a full proposal and costing. We were selected from a group of three final contenders to deliver the solution.

The old, un-scaleable existing PBX was removed and sold to a technology refurbisher. It was replaced with a trio of Intel-based PC servers running 2 Xeon 1Ghz processors each, with 4Gb of RAM and RAID mirrored 80Gb SAS disks, and three network cards per system. A 5KVa “smart” UPS was installed to protect the PBX systems and related gear.

Each server was installed with Linux CentOSⁱ and a current version of Digium’s Asterisk Free Open Source Software (FOSS) PBXⁱⁱ. The first two servers were configured with Linux high-availability “heartbeat” servicesⁱⁱⁱ to allow sub-six-second fail-over times. One machine was the “live” PBX, and the other was the “hot standby”. The third machine was loaded with a FOSS virtual FAX modem stack called IAXModem^iv and a FOSS FAX server software package called HylaFAX^v.

The Customer converted all incoming copper pairs from France Telecom to a pair of 32-channel voice E1s from Orange Business Systems^vi. A third data-only E1 was also brought in and terminated on a fourth server identical to the PBXs, only sporting four network cards and running a Linux IPTables^vii firewall.

The voice E1s were terminated on a Redfone^viii foneBRIDGE ™ Time Division Multiplexing Over Ethernet (TDMoE) device. This device was dynamically reprogrammed to direct the TDMoE stream to which every PBX was the “Live”. Since it is a solid-state unit with no moving parts and its software load is externally managed by the “Live” PBX, it is extraordinarily reliable.

The Customer sourced an Internet Telephony Service Provider (ITSP) with pan-European termination and origination, called WideVOIP^ix. This allowed them to add new phone numbers in four other countries as part of the project without substantial cost. These numbers terminated directly via SIP VoIP onto the Asterisk PBX stack. In addition, any long distance calls placed were sent via this connection whenever possible to avoid international long-distance calling charges.

One network interface card (NIC) was on the data LAN, the second was on the VoIP LAN, and the third was on the TDMoE LAN. This ensured the best possible bandwidth and NIC processor availability for the VoIP environment.

This “Hybrid” configuration of digital E1 and VOIP allowed the greatest flexibility, cost control and reliability.

Each screen had a “call for help” link which triggered an alarm on our system. We could then either call, instant message (IM) or email to contact the user who triggered the alarm, depending on their indicated urgency of issue.

The project, as delivered in 2007, was valued at roughly $100,000 CAD. At the time, The Customer indicated to us this was ½ the next highest quote and neither other competitor could deliver the feature-set we were promising. They realized a Return on Investment (ROI) of 49 months, simply by the dramatic reduction in long-distance fees they were paying to France Telecom; their bill was nearly cut in half. In addition, as noted above in “3.” on page 2, they were able to substantially increase their marketing reach by offering local contact numbers in four other countries with no increase in cost. This would have been unthinkable with a traditional solution.

By using VoIP to TDM converters in the project, The Customer was able to retain all the benefits of both types of telephony. This “Hybrid” system gave all the robustness and call quality normally associated with local TDM calling, but allowed the creation of a custom application for managing the Front Desk work-flow that integrated the telephony management and billing into the process with ease. In addition, more services and features were added, increasing the over-all business value of the phone system, yet keeping capital expenditures well within budget.

The project was considered a substantial success by The Customer. They stated repeatedly that they were both pleased and excited by both the achievement of the desired results, as well as the possibility for future features and services to their clients.

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John usually talks to audiences who are very unhappy because they don’t own the network nor have control over QoS, but WISPs and other ISPs attending ISPCON are in control and do own the network. The voice they sell can be as good or bad as they are willing to enable with the power they have.

Asterisk is a viable tool to provide voIP. Often used as a telephony toolkit, as voie application, set of APIs, and PBX. Decide to make the call go over SIP or PRI or SS7 link. The APIs give call control over other interfaces for web 2.0 apps, pre-existing database to make calls out… The PBX is very popular worldwide.

“You’re using Asterisk. You just don’t know it.”

Asterisk experienced more than a million downloads in 2008. Watch the video for more info and stats. This video was recorded by me Suzanne during ISPcon 2008.

If you the ISP or WISP is not using and selling Asterisk-based product, who is? That’s a question Todd posed. I would say, “Someone is. No, many are.”